sr-users March 2015

sr-users@lists.kamailio.org

88 participants
149 discussions

Kamailio - Asterisk, problem with SIP trunk mode
by Mickael MONSIEUR 02 Dec '15

02 Dec '15

Hello, First, i have 7 years experience with Asterisk, but I started a project with Kamailio, forgive me in advance if I say silly things...! ;-) I set up a classic Asterisk / Kamailio configuration: sip phones -> kamailio -> asterisk -> sip trunks/pstn. When a call comes from the PSTN side, if I configure Asterisk as follows: [012345678] type = friend username = 012345678 secret = xxxxxxx host = dynamic fromdomain = sip.mydomain.com fromuser = 012345678 Standard mode: exten => 012345678, 1, Dial(SIP/012345678) -> The call is redirected on the phone by Kamailio ! :-) ------------------------------------------------------------------------------------------------------------------------------------------------ Trunk mode: [mytrunk] type = friend username = mytrunkUser secret = xxxxxxx host = dynamic fromdomain = sip.mydomain.com fromuser = mytrunkUser exten => 012345678, 1, Dial(SIP/mytrunk/012345678) -> The call is rejected by Kamailio.... exten => 012345679, 1, Dial(SIP/mytrunk/012345679) -> The call is rejected by Kamailio .... My question is how to allow the routing of multiple numbers (trunk mode) in a SIP account with Kamailio? Best regards, Mickael

3 6

Kamailio IMS - VM
by Bruno Cruz 17 Nov '15

17 Nov '15

Hello everyone, My name is Bruno, I'm a student from Portugal studying IMS at the moment. I've been following tutorials online, trying to get Kamailio to work as an IMS core - I'm pretty much a newbie in this field. Unfortunantly I've been getting several errors and I can't seem to properly configure everything. I was wondering if there is a All-in-One VM that can be downloaded, with Kamailio ready to be used as an IMS core? Thanks in advance. Best regards, Bruno C.

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Interpretation of Contact header
by Joachim Büchse 30 Jul '15

30 Jul '15

Good day, I’m experiencing some problems with our VoiP providers handling of REGISTER requests. We are using a Gigaset PRO N720 as UAC behind a Juniper SSG 140 with SIP-Alg enabled. This setup kind of works with UDP but our provider wants us to use TCP. With TCP enforced incoming calls don’t work. I’ve done some wire tracing and to me it seems that the providers configuration is to blame, but then - there are many RFCs out there and many NAT and UAC bug workarounds. Anyway, I wanted to get the opinion of “the" experts about how the requests send to the UAS SHOULD be correctly interpreted. The REGISTER requests/responses look like this (outside of the firewall): Protocol TCP! client port 19091 <-> server port 5060 REGISTER sip:pbx.peoplefone.ch SIP/2.0 Via: SIP/2.0/TCP 212.126.160.92:6717;rport;branch=z9hG4bKc1375589832468de63a719eac31156ec From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch> Call-ID: 2825358480@10_10_128_10 CSeq: 1 REGISTER Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp> Max-Forwards: 70 User-Agent: N720-DM-PRO/70.089.00.000.000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 212.126.160.92:6717;rport=19091;branch=z9hG4bKc1375589832468de63a719eac31156ec From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=a0440f545f39b2694d387b475a5f6bc9.b8fc Call-ID: 2825358480@10_10_128_10 CSeq: 1 REGISTER WWW-Authenticate: Digest realm="pbx.peoplefone.ch", nonce="VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy" Server: kamailio (3.2.1 (x86_64/linux)) Content-Length: 0 REGISTER sip:pbx.peoplefone.ch SIP/2.0 Via: SIP/2.0/TCP 212.126.160.92:6717;rport;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0 From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch> Call-ID: 2825358480@10_10_128_10 CSeq: 2 REGISTER Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp> Authorization: Digest username="90780408050", realm="pbx.peoplefone.ch", uri="sip:pbx.peoplefone.ch", nonce="VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy", response="764f371a08d258157a249f8d1b852514" Max-Forwards: 70 User-Agent: N720-DM-PRO/70.089.00.000.000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/TCP 212.126.160.92:6717;rport=19091;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0 From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485 To: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=a0440f545f39b2694d387b475a5f6bc9.6bda Call-ID: 2825358480@10_10_128_10 CSeq: 2 REGISTER Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp>;q=0;expires=180;received="sip:212.126.160.92:19091;transport=TCP" Server: kamailio (3.2.1 (x86_64/linux)) Content-Length: 0 The ip:port the firewall is sending those requests from is ip 212.126.160.92 port 19091. So this does NOT match the port from the Contact header. For TCP this seems rather logical to me, as one cant be listening on a TCP port and use it for sending at the same time. The UAC closes this “register connection” with TCP FIN after the register, and so does the firewall. However unfortunately subsequent requests from the provider (ie UAS) come in on port 19091 (not port 6717 from the Contact header) and the firewall simply drops them. Observations: - the server does NOT include received=212.126.160.92 in the Via of the reponse. According to RFC3581 this is mandatory when rport is present in the request, so this is probably an error in the server. - the server does include received="sip:212.126.160.92:19091;transport=TCP” in the Contact of the response. I didnt see this in any RFC (which means nothing;-) but it could be an error. - after the client received the 200 OK it closes the TCP connection. - the server tries several times to re-contact the client (incoming TCP SYN). However not on port 6717 (defined in the Contact header) but on port 19091 (where the REGISTER came from). RFC3581 defines special behaviour when “rport” is defined in the request (i.e. response should go to the same port the request came from) - however it’s not so clear if this should apply to subsequent (INVITE/OPTIONS) requests from the server to the client. Those are strictly spoken not replies (or are they?). RFC5626 defines that a “proxy” should keep track of the flows over which it received a registration and send requests over the same flow. It is not clear if RFC5626 should be applied. The RFC5626 defines that a UAC includes an “ob” parameter in the Contact field if it whishes further requests over the same flow. Also the RFC mandates a client to add a "reg-id=x" in the Contact field. Both are not the case here, so in short I think RFC5626 should NOT be applied. In which case conecting to the originating port (instead of the Contact port) would be a server error. So in short and if I interpret the RFCs correctly, the client is reachable and should be contacted on Transport: TCP IP: 212.126.160.92 Port: 6717 If anyone who lives and breathes SIP could enlighten me if the UAS is right to call back on 19091 instead of 6717 I would really appreciate it;-)) Best regards, Joachim

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[Kamailio-Users] HA+DRBD setup
by Siddhardha Garige 09 Jul '15

09 Jul '15

Hello all, Just wondering if anyone know any tutorial on setting up HA+DRBD solution for kamailio. Especially creating partitions, DRBD devices and mount points. Thanks in advance, -Sid "May the light be with you." ______________________________________________ Siddhardha Garige www.luminepixels.com

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topoh ACK Call-ID mismatch
by Daniel Tryba 23 Apr '15

23 Apr '15

Trying to call a T.38 enabled endpoint B from an endpoint A that doesn't. This results in a "488 Not Acceptable" from A. The kamailio (4.0.3) in between with topoh enabled (with mask_callid set to 1. Kamailio ACKs the 488 to A, but the ACK has the wrong (masked) Call-ID, resulting in the ACK to A being ignored, A keeps sending 488 at intervals before just dropping the call. My understanding is the Call-ID in the ACK has to be the same as the Call-ID from the 488. But instead of sending the unmasked Call-ID used in all other messages between A and Kamailio, the ACK contains the masked Call-ID used in all messagea between A and Kamailio. I can't find a bug report relating to this. Is this a known feature or maybe fixed in newer versions? A->Kamailio SIP/2.0 488 Not acceptable here. Via: SIP/2.0/UDP 109.235.32.40;branch=z9hG4bKf91a.c6c88747.0;received=109.235.32.40. Via: SIP/2.0/UDP 172.19.162.1;branch=pocos- rS4MusXox1l5QHyNxRy6uAXsEOdsxidSEAktxGZKWgeKCgeS- RrKEAy057Nl9DEpWpefZD6AhGtwWDJjEqeuEAZgZs5iZA7*. From: <sip:+31880100799@sip.pocos.nl:5060>;tag=as3869fe2a. To: "+31880100705" <sip:+31880100705@sip.pocos.nl>;tag=as1b0b8097. Call-ID: 0b5946b977210450571f767a19cd6fa0(a)99sip.pocos.nl. CSeq: 102 INVITE. Kamailio<-A ACK sip:+31880100705@109.235.32.48 SIP/2.0. Via: SIP/2.0/UDP 109.235.32.40;branch=z9hG4bKf91a.c6c88747.0. Max-Forwards: 16. From: <sip:+31880100799@sip.pocos.nl:5060>;tag=as3869fe2a. To: "+31880100705" <sip:+31880100705@sip.pocos.nl>;tag=as1b0b8097. Call-ID: !!:xqXtWRMpZAngEsX3xGMtxGrgxDZgEA9JxR4AzGz8ZRIyWR4sh.yomqlACgxoC8K*. CSeq: 102 ACK. -- Telefoon: 088 0100 700 Sales: sales(a)pocos.nl | Service: servicedesk(a)pocos.nl http://www.pocos.nl/ | Croy 9c, 5653 LC Eindhoven | Kamer van Koophandel 17097024

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Kamailio: MI Command: 'trusted_dump' and 'trusted_reload'
by José Luis Millán 22 Apr '15

22 Apr '15

Hi, I'm using 'allow_trusted()' from Permissions module without a problem in a Kamailio 4.0 routing logic. It does just work, but the MI command 'trusted_dump' tells me that the module is not in use: ''' server:/home/jmillan# kamctl fifo trusted_dump 500 Trusted-module not in use ''' Doing a 'trusted_reload' does neither make any query to database to retrieve the data. Other MI commands of same module do work correctly. ie: 'address_reload' or 'address_dump' Note: I have tested in a Kamailio 1.5 and the result is the same. Regards. -- José Luis Millán

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Roadmap for next major release v4.3.0
by Daniel-Constantin Mierla 22 Apr '15

22 Apr '15

Hello, it is time to nail down the roadmap to the next major release. We discussed during the last IRC devel meeting, proposing to get it out by beginning of June. Given we need to have at least one month of testing, I propose the next milestones: - freezing the development: Wednesday, April 22, 2015 - if testing goes smooth, then branching 4.3 after about one month: During the week starting May 18 - test more in beta phase, prepare packaging, etc. and release after 2-3 weeks: One of the days between June 4 and 11 Other suggestions or adjustments are welcome! Send them to mailing list to discuss further. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

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missing BYE when 2 redundant kamailio servers share the same database
by Ding Ma 08 Apr '15

08 Apr '15

Hi, all I'm trying to set up 2 kamailio servers for active-active redundancy. The two kamailio severs share the the same database with db_mode=3, and no registration replication. Use pjsua2 as SIP client for testing. The test setup is as follows: kamailio server 1(k1): 10.0.1.30:5061 <http://10.0.1.30:5061/> kamailio server 2(k2): 10.0.1.32:5061 <http://10.0.1.32:5061/> sip client c1: sip:16317@10.0.1.30 <mailto:sip%3A16317@10.0.1.30>, client ip: 10.0.1.254 sip client c2: sip:72316@10.0.1.30 <mailto:sip%3A72316@10.0.1.30>, client ip: 10.0.1.254 c1 is registered with k1. c2 is registered with k2. When c1 calls c2, the call flow is as follows: INVITE: c1------>k1------>k2------>c2 200 OK: c1<------k1<------k2<------c2 So far, the INVITE works. c1 and c2 can establish a call. However, when c2 hangs up and sends BYE, the BYE is forwarded all the way to k1, and then k1 gives a 500 server error. The log on k1 seems to indicate k1 was attempting to create a TLS connection to itself instead of using the existing TLS connection to c1 to forward the BYE. BYE: c1 X k1<------k2<------c2 | |------->Connection refused/500 Here is the kamailio log on k1: Mar 16 10:34:21 kamailio.sip.com <http://kamailio.sip.com/> /usr/sbin/kamailio[13396]: ERROR: <core> [tcp_main.c:2740]: tcpconn_1st_send(): connect 10.0.1.30:55999 <http://10.0.1.30:55999/> failed (RST) Connection refused Mar 16 10:34:21 kamailio.sip.com <http://kamailio.sip.com/> /usr/sbin/kamailio[13396]: ERROR: <core> [tcp_main.c:2750]: tcpconn_1st_send(): 10.0.1.30:55999 <http://10.0.1.30:55999/>: connect & send for 0x7fc96a68a1a0 failed: Connection refused (111) This is the routing logic for BYE in kamailio.cfg: if (loose_route()) { route(DLGURI); if (is_method("BYE")) { xlog("L_DBG", "=====BYE $ru from $fu $si:$sp to $du=====\n"); dlg_manage(); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } else {...... If I add location based routing for BYE from peer kamailio, then the BYE seems to be forwarded correctly: if (loose_route()) { route(DLGURI); if (is_method("BYE")) { xlog("L_DBG", "=====BYE $ru from $fu $si:$sp to $du=====\n"); dlg_manage(); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails # If BYE coming from Kamailio peer, route BYE by location $var(peerlist)=$sel(cfg_get.trusted.peers); $var(i) = 0; while($var(i)<$(var(peerlist){param.count})) { xlog("L_DBG", "=====$(var(peerlist){param.count})=====$(var(peerlist){param.valueat,$var(i)})=====\n"); if(src_ip==$(var(peerlist){param.valueat,$var(i)})) { lookup("location"); xlog("L_DBG", "=====BYE from $fu $si:$sp to $du=====\n"); break; } $var(i) = $var(i) + 1; } } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } else {...... My first question is why k1 loose_route sends the BYE to itself instead of the client. Is this a bug? My next question is whether the above location routing for BYE from peer kamailio a good/safe approach. The SIP traces will be sent later to avoid exceeding email size limit. Thanks

3 5

Variable interpolation in xavp identifiers
by Alex Balashov 08 Apr '15

08 Apr '15

Hello, There seems to be an implementational difference in interpolation of PVs in AVP and XAVP identifiers and/or field names. The pseudovariables documentation[1] on classical AVPs says: The 'id' can be: ... pseudo variable - if value of pv is integer, id is integer, if string, id is string So, for example, this composition of variables works: $var(f) = 'donkey'; $avp($var(f) = 'eehaw'; xlog("L_INFO", "Donkey says: $avp(donkey)\n"); It correctly prints: Donkey says: eehaw Making some assumptions, I tried this, and it did not work similarly as expected: $var(f) = 'donkey'; $xavp(r=>$var(f)) = 'eehaw'; xlog("L_INFO", "Donkey says: $xavp(r=>donkey)\n"); This prints: Donkey says: <null> And a PV dump confirms that the PV value is not interpolated: Mar 31 18:26:15 localhost /usr/local/sbin/kamailio[2399]: INFO: debugger [debugger_api.c:1799]: dbg_dump_json(): {"$xavp(r)":[{"$var(f)":["eehaw"]}]} Interpolation of variables does not work for either field or stem names. Is this a design principle of XAVPs, or an oversight? Thanks, -- Alex [1] http://www.kamailio.org/wiki/cookbooks/4.2.x/pseudovariables#avp_id_-_avps -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

Websockets, set_contact_alias(), & IPv6
by Anthony Messina 08 Apr '15

08 Apr '15

I've been working on Kamailio websocket integration and I believe I'm having issues with the IPv6 address representation in the Contact header's alias parameter. After Googling, it appears after https://github.com/kamailio/kamailio/commit/814c08f3 the IPv6 contact is represented in brackets []. However, it when using websockets, and the alias= parameter uses brackets [] around and IPv6 address, there are message parsing issues. With a header such as the following Contact: <sip:wstest1@example.com;gr=urn:uuid:26140e27-0ab7-4e65-98e3-3d0909b1434e;alias=[2001:db8:0:1]~48768~6> Asterisk 13.2.0 will give the following error: pjsip:0 <?>: sip_transport. Error processing 1855 bytes packet from UDP 10.77.79.3:5060 : PJSIP syntax error exception when parsing 'Request Line' header on line 12 col 129: And when sipjs, or jssip are used with either Firefox or Chrome, they send garbage in the ACK request URI: Kamailio logs something like the following and the ACK cannot be processed: WARNING: sanity [sanity.c:236]: check_ruri_scheme(): failed to parse request uri [�a�{1me▒s�na@50�9��1.�8:�2v0;i��a��=11>7�n7x>10O�2v0~1] Is it proper to have [] brackets around the IPv6 alias address in the Contact header? Does the value need to be quoted? If I force the browser to use IPv4, without changing anything else, both jssip and sipjs work perfectly in Firefox and Chrome. -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

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sr-users March 2015